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Error 503 at startup
by William Colls 28 Apr '17

28 Apr '17

FreeSWITCH version 1.6.12 Debian Jessie 64 bit When the systen boots up, freesitch is started, but it does not connect to my voip provider. The following messages are reported in fs_cli: [NOTICE] sofia_reg.c:448 Registering voipms_demo [ERR} sofia_reg.c:2446 voipms_demo Failed Registration with status DNS Error [503] If I do a systemctl restart freeswitch.service, the connection is made without problem. Should I be talking to my voip provider about this, or is it something miss-configured within FreeSWitch? Thanks for your time William.

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Reply of reinvite of 3pcc call with more than one codec
by José Lopes 28 Apr '17

28 Apr '17

Hello, I have a scenario that Freeswitch receive a call without SDP (3pcc call) and this is ok. But when Freeswitch receive a re-invite on that 3pcc call, it replies with 200 OK but with more that one codec. I notice that when Freeswitch receive a call with SDP and after that, it receives a re-invite, it replies with only one codec. The reply of the re-invite of 3pcc call with more than one codec causes trouble on the other endpoint. It is possible to configure or change this behavior of Freeswitch on 3pcc call? Best regards, José Lopes

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'Tapping' of T38 Gateway call
by Raúl Alexis Betancor Santana 28 Apr '17

28 Apr '17

Hi all, I'm working with a FS 1.6.17 server, acting as T38 endpoint to a HylaFax server and also as T38Gateway for some T.38 SIP Endpoints. I wonder if could be possible that on the T38Gateway calls, I could save the pages that are going throught my FS server. The a-leg reaching the FS server could be uLaw FAX or T38 Fax, doesn't matter ... the b-leg going from the FS sever to my internal T38 endpoints are allways T38, thats why I used T38Gateway App. Is there any way to write down to the FS disk the pages been transfered ? Best regards

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No voice when using GSMOpen
by Raymond 28 Apr '17

28 Apr '17

Hi all, I'm using GSMOpen module with freeswitch master version (pulled from git). The usb dongle for test is Huawei E173 , hardware info checked by dc-unlocker is : Found modem : E173 Model : Huawei E173 IMEI : 860872014****** Serial NR. : ADABYA9340****** Firmware : 11.126.25.00.76 Compile date / time : Mar 27 2012 15:57:50 Hardware ver. : CD1E153M Dashboard version : UTPS23.009.09.00.76_MAC23.009.09.00.76 Chipset : Qualcomm MSM6290 Voice feature : enabled SIM Lock status : unlocked Wrong codes entered : 0 (unlock attempts left : 10) I've make test 2 ways. Test A: Using another phone call-in Freeswitch ,with IVR module , caller will hear the ivr voice , dtmf function worked well ,but record file have only called party voice ( ivr voice), caller have no voice record (i'm sure the dialplan record function workwell when NOT using gsmopen module). Test B: Freeswitch+gsmopen as call out. I'm using MicroSIP softphone make a sip call ,and gsmopen as b-leg to deliver the call. ( <actionapplication="bridge" data="gsmopen/ANY/${destination}"/> ) I tried TestB several times , the best result , called party hear the caller voice , but caller cann't hear called party voice always. So ,Any advice to resolve the problem will be appreciate. Raymond BTW , And I monitor the diagnostic port, when usb dongle got an incoming call. got following: ^ORIG:1,0 ^CONF:1 ^RSSI:22 ^CEND:1,0,29,16 ^MODE:3,3 ^RSSI:19 ^RSSI:24 ^MODE:3,3 +CREG: 1 +CGREG: 1 RING +CLIP: "1335571xxxx",161,,,,0 // 1335571xxxx is my another chinese cellphone number make the call. ^RSSI:21 ^CONN:1,0 ^MODE:3,2 +CREG: 1 +CGREG: 1 ^CEND:1,7,104,16 ^MODE:3,3 ^RSSI:24 ^MODE:3,3 +CREG: 1 +CGREG: 1

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fs deadlock
by Byron Zhang 张永锋 27 Apr '17

27 Apr '17

Hi I have a problem, when i use app sofia_contact via esl, if CPS is a little bigger, fs will deadlock. Has anyone encountered the same problem? Is there any way to replace app sofia_contact? The version of my fs is 1.2.3. Thanks

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fs deadlock
by Byron Zhang 张永锋 27 Apr '17

27 Apr '17

Hi Outstanding issues to be helped: 1. When using esl to call Freeswitch's app(sofia_contact), cps was between 1 to 2, Freeswitch was leading to deadlock. 2. Also, when using lua to manage SIP register instead of xml, when Sofia_contact was using in Dial-striing, it also led to Freeswitch deadlock. You can search "loac_wait" in the attached Freeswitch thread to find deadlock Null. Thanks a lot!

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Handling Variables With Commas
by Markus Bönke 26 Apr '17

26 Apr '17

Hello, I try to set a sip_header to a comma separated value. If I set a X-Header like: {sip_h_X-My-Header=one\,two\,three,leg_time_out=10,process_cdr=b_only} as described on the wiki, it works. But if I try to set a regular Header like: {sip_h_Privacy=header\,user\,id} the values are separated by semicolon instead of comma Privacy: header;user;id Is this by design? Is it possible to set comma separated values on regular headers? Thanks and regards Markus

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Fwd: zrtp_enrollment appears broken since 1.6.13
by Vydriduch Vychcany 26 Apr '17

26 Apr '17

Hello I have been happily using ZRTP in a trusted MITM setup running FS 1.6.12 on Ubuntu 16.04.2 LTS (Debian Jessie packages) with Linphone clients. If I upgrade to a newer (>=1.6.13) stable version, ZRTP encryption no longer works. I have attached what FS 1.6.17 and the client say about ZRTP in the log files. Apparently, FS appears to receive no ZRTP handshake traffic at all and then simply gives up. Before I get lost in the source code, can anyone please confirm that zrtp_enrollment actually works in recent stable releases? Regards Vydriduch

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intermittent build error
by Dave Horton 26 Apr '17

26 Apr '17

I am trying build 1.6 from source and intermittently getting the following error: # wget -O - https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add - converted 'https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub' (ANSI_X3.4-1968) -> 'https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub' (UTF-8) --2017-04-26 18:25:04-- https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub Resolving files.freeswitch.org (files.freeswitch.org) 209.105.235.7, 2607:f348:1021::7 Connecting to files.freeswitch.org (files.freeswitch.org)|209.105.235.7|:443... connected. GnuTLS: A TLS warning alert has been received. GnuTLS: received alert [112]: The server name sent was not recognized ERROR: The certificate of 'files.freeswitch.org' is not trusted. ERROR: The certificate of 'files.freeswitch.org' hasn't got a known issuer. Any ideas?

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Executive/Assistant Feature
by SamyGo 26 Apr '17

26 Apr '17

Hi All, We are developing Executive/Admin feature and I was wondering if anyone has done something similar in the past to help us out. We require Barge to work for multiple shared line appearances. These will be assistants and their executives using SCA. Assistants should be able to Barge exec line and setup a server-hosted (managed) conference. So, I can imagine that at anytime barge is done it converts/transfer the call to a conference room. Management of conference will be done via a web page. Our goal is to use minimum channels on the phone when barge + conference is used allowing assistant to take new incoming calls and setup conferences for multiple execs at the same time. We're not expecting anymore than 3~5 ongoing conferences handled by the assistant. Thanks, Sammy

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